Tagged Questions

Fiter design is the process of designing a filter.

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3answers
105 views

Audio EQ Cookbook without frequency warping

The famous http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt offers a set of [real] biquad filter calculation formulas that generally work fine. However, when filter's frequency approaches Nyquist ...
1
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0answers
22 views

filter design for ECG

I would like to ask how to design a digital filter to remove $50$ Hz interference on an ECG signal. Do I need to use notch filter? Thanks.
3
votes
5answers
86 views

What's the Q factor of a digital filter's pole?

For the analog S-plane, the Q factor depends on the angle of the complex pole $p$ from the horizontal axis, $\theta = \arg p$: damping factor $\zeta = \frac{1}{2Q}$, and $\zeta = \cos(\theta)$, and ...
0
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0answers
19 views

Basic Fourier Transform filtering and LTI frequency response using Matlab

Not sure if this is the right place to ask... I'm new to Matlab for LTI signal processing and wondering if anyone can help with something that I'm sure is meant to be basic. I've spent hours and hours ...
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0answers
41 views

Is it possible from 1 order cascading low pass filters to make a butterworth filter

Today I have heard, that is "impossible" to make low pass butterworth filter from cascading 1 order low pass filters(RC). I am trying to discover is that true, but I can find little to none ...
1
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1answer
40 views

Pole/Zero Placement for Filter

I'm attempting to design a FIR high pass filter than keeps signals above 200Hz and rejects signals below 60Hz with a sampling frequency of 500 samples/sec. This is my first time attempting this and ...
0
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2answers
35 views

How do I take the real part of this bandpass filter system's output?

I'm stuck on a final step in this problem. Essentially, there are the two systems above, which we'll call System 1 (Fig. 4.26, with ideal lowpass $H(jw)$) and System 2 (with $H_1(jw)$). The ...
0
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1answer
46 views

Identify the system from input and output

I am using Excel. If I have an input and an output, and I know that a recursive band-pass filter produced the output, how can I can find the coefficients of that filter?
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2answers
28 views

Filter design / Analysis apps

There are a plethora of tools, both commercial and free, which I have found online for designing filters. The ones I have tried (so far) prompt for frequency response, number of steps (for FIR), then ...
1
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0answers
27 views

Which values do I use to filter a specific frequency with an RLC circuit?

I have a RLC circuit with transformation $$H(s) = \frac{R}{Ls+\frac{1}{Cs}+R}$$ and I know that by setting $s=j\omega$ I can obtain the frequency response of this system. What I don't understand is ...
0
votes
1answer
48 views

What is the Roll-Off factor of a Root Raised Cosine Filter?

When designing a root raised cosine filter on Matlab, there is an option to set its "roll-off factor". What does this mean? I have been looking around for a while but I can't seem to find a clear ...
0
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0answers
27 views

FIR filter design using windowing

I am a newbee in DSP. I am using an FIR low pass and high pass filter with rectangular window in C++. I have calculated the filter coefficients using Low pass filter kernel: High pass filter ...
0
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2answers
66 views

FIR vs IIR filter

In my filter design, FIR gives order 14 and IIR gives order 8 or 4. so what are the disadvantages of IIR design?
0
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1answer
45 views

Why bother impluse invariance, bilinear transform?

Now that we can operate in the frequency domain by way of FFT. We can implement ideal lowpass, highpass and bandpass digital filters. then why there many textbooks talk about designing digital ...
0
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2answers
51 views

multiplications per second for FIR filter

In my filter the fs=300MHz. and no. of coefficients is 31. I have following queries. (1) what will be the number of multiplications per second? (2) can multiplication be done in one clock cycle? (3) ...
0
votes
1answer
54 views

FIR filter design

I am designing an FIR filter.my specs are fs=300MHz, Fc=45MHz, Fs=75MHz, passband gain=3dB , stopband attn=>40dB. what parameters I values of a have to provide in firpm for the minimum order filter ...
0
votes
2answers
49 views

How to pad a windowed sinc filter in the frequency domain

I'm creating windowed sinc filters to apply them to certain signals that I'm dealing with. To design the filters, I'm using the approach described in the book "The Scientist and Engineer's Guide to ...
0
votes
1answer
38 views

How to periodically estimate states of a LTI if the output is measured irregularly?

How can I periodically estimate the states of a discrete linear time-invariant system in the form $$\dot{\vec{x}}=\textbf{A}\vec{x}+\textbf{B}\vec{u}$$ $$\vec{y}=\textbf{C}\vec{x}+\textbf{D}\vec{u} ...
0
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1answer
26 views

High Speed Digital Signal Processors [closed]

Can anybody suggest me the highest speed DSPs available in the market now. Also, what would be the best for a filer design application a DSP or an FPGA?
0
votes
1answer
60 views

group delay of the FIR filter

For an FIR filter, with symmetrical tap values $h[N-1-n]=h[n]$, why is the group delay $\frac{N-1}{2} T$ (where $N$ is the number of taps of the FIR filter and $T$ is the sampling time)? Why is ...
0
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2answers
72 views

direct Frequency domain FIR filtering vs Overlap-add method

I'm trying to do bandpass filtering of a EEG signal samples at 250Hz and benchmarking the following 4 methods of FIR filtering for different filter orders. The length of the signal is 15000 samples. ...
0
votes
2answers
61 views

why the number of filter coefficients in FIR filter has to be an odd number?

I calculated the order of the FIR filter to be 31(N). so, the number of coefficients has to be 32 (N+1). so, I have to increase it to 33 to make it an odd number. Why the number of filter ...
-1
votes
1answer
34 views

Accumulator with decay?

I am using an accumulator in the form of X + [Y-1], X plus the unit delay of the output. Is there an accumulator that follows this form, but will fall to 0 over time, similar to a differentiator?
0
votes
1answer
96 views

Picking out a signal that appears to be noise inside a large signal

I have a signal from a photodiode sensor that has two types of noise. One type of noise is ambient light white noise that gets introduced just from the surrounding environemnt. The other type of noise ...
1
vote
1answer
54 views

Creating a digital filter, from laplace to z transformation(zero order hold) to code?

I'm trying to create a digital filter in code(C) but any languge is fine. Now I've got an analogue filter that i have represented by an equation in the Laplace domain and I want to try and implement ...
0
votes
1answer
31 views

Implementing digital filter by padding with zeros

I am making a 9th order digital bandpass filter with lower and upper corners of 200 kHz and 40 MHz respectively. I am using this filter to filter a 1D time domain signal which is 64k samples long ...
2
votes
2answers
79 views

Reconstruct FIR frequency response from the coefficients

There is a lot of tools for designing FIR filters. Using one such tool I have designed a filter that satisfies my needs. It's coefficients are as follows: ...
1
vote
1answer
51 views

correct for shift from butterworth filter?

I have implemented a python butterworth filter but it seems to shift the data a lot, the x location of my data is very important to me for my analysis. Is there a way to correct for this shift ...
1
vote
1answer
239 views

Real and Imaginary Frequency Responses of a single complex pole

I'm filtering a real signal with a single complex pole with a complex coefficient (a_re [real part] and a_im [imaginary part]), I also have a gain coefficient but I'm gonna leave it out for the sake ...
0
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0answers
41 views

Using Python with scipy.signal firwin, how do I know how many taps/coefficients should be used?

I need to alter an FIR filter which I know very little about. I have a signal which need to be altered, and looking at a Python script used for this task, I see that it uses Scipy's signal library, ...
-1
votes
1answer
50 views

Slepian or DPSS window

I am planning to use Slepian or DPSS window in my application where I want central lobe to be concentrated and also have low bandwidth: ...
2
votes
3answers
81 views

How to filter noise from continuous path in image

Problem statement In the dataset above, you can notice a continuous path and then a noise-band at around 320-400 on the x-axis. I want to be able to apply a filter that amplifies the continuous ...
-1
votes
1answer
66 views

Low pass filter to maintain edge information

I am looking for a kernel acting as a low pass filter that satisfies these conditions: $$K(-\mathbf{u})=K(\mathbf{u}) \tag{1}$$ $$K(\mathbf{u}) \ge K(\mathbf{v}), \;\; \text{if} \;\; ...
0
votes
1answer
60 views

Design and implementation of causal band-pass filter for biosignals - what to consider? [closed]

I currently work in a project of clinical software (C#) which deals with clinical biosignals, specifically EMG and other low-frequency signals (load-cells, goniometers, and other biomechanical ...
0
votes
1answer
48 views

Approximate a system frequency response with a filter in Matlab

suppose I know the frequency response of a (linear) model approximating a real physical system but only at a specific frequency $f_0$ (so basically I have a complex number whose module is the ...
0
votes
0answers
59 views

Any way to compensate for fractional delay filter's weakness between taps?

A little background: I have an FPGA (running at 200MHz) that I am creating multipath events using buffers, all of that works. I take RF, mix down to IF, sample, make adjustments, out the DAC, and ...
0
votes
2answers
75 views

Low-pass vs median filter delay

I need to design a filter to reduce noise from accelerometer readings and use this data to a control system. One of the problems of a simple low-pass filter (like a single pole IIR filter) is the ...
1
vote
1answer
51 views

Designing the filtering for a time domain data acquisition application

I want to filter sensor signals for a data acquisition application on an ATmega microcontroller. The interesting parts of the signal are in the range 0-5Hz, and I'd like to leave these mostly ...
0
votes
1answer
85 views

Best way to inverse Hilbert on FPGA?

I have a real signal that is being sampled via ADC and passed to a Xilinx V6 FPGA. I then do a Hilbert transform on it using the FIR compiler core and get the complex signal (I and Q) to tweak the ...
0
votes
4answers
109 views

Why are recursives methods useful for FIR filter design?

As I understand it, FIR filtering is a linear process. That mean for me that the whole filtering process will have a fully predictable behavior. So, could someone explain why a universal ...
3
votes
1answer
84 views

Confusion about result of FIR All-pass filter design

I try to design a FIR All-pass filter with random phase in the frequency domain. I am a bit confused by my result and am not sure if the reason is a programming error or a misconception about ...
0
votes
1answer
41 views

Is there a way to break an analog sound signal up and send it through two radio chips at double the data rate of one?

Here is a radio chip for audio that sends at a CD-level data rate. Could you sample an analog signal (for instance from a guitar pick-up or microphone) in such a way that you could transmit using two ...
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votes
1answer
50 views

Applying Discrete Filters

I have an exercise in DSP. I learnt about discrete filter but I don't know how to apply it to produce an output from an input signal. Please guide me to do this: Assume fundamental sample time is ...
2
votes
2answers
275 views

How do I manually plot the frequency response of a bandpass Butterworth filter in MATLAB?

I have code like below that applies a bandpass filter onto a signal. I am quite a noob at DSP and I want to understand what is going on behind the scenes before I proceed. To do this, I want to know ...
0
votes
1answer
113 views

FIR-Decimation and Low-pass filter (taps vs number of input points vs number of decimation stages)

I am new to digital filters and trying to understand parameters behind them. Scenario I am trying to get DC value from signals sampled at 100kHz sampling rate. such high frequency is used due to ...
0
votes
0answers
22 views

IIR Design with Neural Networks

i'm trying to apply this paper: using C++ Design of Digital Recursive Filter Using Artificial Neural Network In this paper the weights are the coefficients and they start with weight a(0) = 1 and ...
0
votes
1answer
83 views

First-order low pass filter with bounded derivative

Given a function $x(t)$ and a first order low-pass filter $\frac{dy}{dt} = \frac{1}{\alpha} (x(t) - y(t))$, is there a way to choose $\alpha$ analytically such that the derivative of $y(t)$ is ...
2
votes
1answer
95 views

How to remove historical values from exponential filter

Suppose I have two time series $A:=[0,0,0,4,5,6]$ and $B:=[1,2,3,4,5,6]$. I implement the following filter (with initial value at the first element): $$ f(t) = f(t-1) + \alpha*(f(t)-f(t-1))$$ with ...
1
vote
2answers
208 views

Low Pass Filter in AM Demodulation

I am trying to recover my baseband signal that was AM modulated. I am using the filter command in MATLAB to perform the low pass filtering ...the code is given below: ...
1
vote
2answers
139 views

Difference between Equiripple & Least Squares design for FIR digital filters

For an efficient and optimized digital FIR filter design, there are two methods available broadly, Equiripple filter design & Least Squares filter design. A general method for designing a filter ...